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Audio: Volume: Add passthrough functions
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Add passthrough functions implementation. If the gain of all
channels equal 0dB, then we use passthrough functions to process
the volume component.

Signed-off-by: Andrula Song <[email protected]>
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@andrula-song
andrula-song committed Jun 21, 2023
1 parent d277d92 commit fc5e117
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Showing 9 changed files with 1,061 additions and 44 deletions.
49 changes: 46 additions & 3 deletions src/audio/module_adapter/module/volume/volume.c
View file
Original file line number Diff line number Diff line change
Expand Up @@ -267,8 +267,10 @@ static inline int32_t volume_windows_fade_ramp(struct vol_data *cd, int32_t ramp
*/

/* Note: Using inline saves 0.4 MCPS */
static inline void volume_ramp(struct vol_data *cd)
static inline void volume_ramp(struct processing_module *mod)
{
struct vol_data *cd = module_get_private_data(mod);
struct comp_dev *dev = mod->dev;
int32_t new_vol;
int32_t tvolume;
int32_t volume;
Expand Down Expand Up @@ -326,6 +328,29 @@ static inline void volume_ramp(struct vol_data *cd)
}
cd->volume[i] = new_vol;
}

cd->is_passthrough = cd->ramp_finished;
for (i = 0; i < cd->channels; i++) {
if (cd->volume[i] != VOL_ZERO_DB) {
cd->is_passthrough = false;
break;
}
}

#if CONFIG_IPC_MAJOR_4
cd->scale_vol = vol_get_processing_function(dev, cd);
#else
struct comp_buffer *sourceb;
struct comp_buffer __sparse_cache *source_c;

sourceb = list_first_item(&dev->bsource_list,
struct comp_buffer, sink_list);
source_c = buffer_acquire(sourceb);

cd->scale_vol = vol_get_processing_function(dev, source_c, cd);

buffer_release(source_c);
#endif
}

/**
Expand Down Expand Up @@ -368,6 +393,7 @@ static void reset_state(struct vol_data *cd)
cd->vol_ramp_elapsed_frames = 0;
cd->sample_rate_inv = 0;
cd->copy_gain = true;
cd->is_passthrough = false;
}

#if CONFIG_IPC_MAJOR_3
Expand Down Expand Up @@ -397,6 +423,7 @@ static int volume_init(struct processing_module *mod)
}

md->private = cd;
cd->is_passthrough = false;

/* Set the default volumes. If IPC sets min_value or max_value to
* not-zero, use them. Otherwise set to internal limits and notify
Expand Down Expand Up @@ -605,6 +632,7 @@ static int volume_init(struct processing_module *mod)
cd->mailbox_offset += instance_id * sizeof(struct ipc4_peak_volume_regs);

cd->attenuation = 0;
cd->is_passthrough = false;

reset_state(cd);

Expand Down Expand Up @@ -971,6 +999,17 @@ static int volume_set_volume(struct processing_module *mod, const uint8_t *data,
cd->ramp_finished = false;
}

cd->is_passthrough = cd->ramp_finished;

for (i = 0; i < channels_count; i++) {
if (cd->volume[i] != VOL_ZERO_DB) {
cd->is_passthrough = false;
break;
}
}

cd->scale_vol = vol_get_processing_function(dev, cd);

prepare_ramp(dev, cd);

return 0;
Expand Down Expand Up @@ -1161,7 +1200,7 @@ static int volume_process(struct processing_module *mod,
}

if (!cd->ramp_finished) {
volume_ramp(cd);
volume_ramp(mod);
cd->vol_ramp_elapsed_frames += frames;
}

Expand Down Expand Up @@ -1291,8 +1330,12 @@ static int volume_prepare(struct processing_module *mod,
ret = -ENOMEM;
goto err;
}
#if CONFIG_IPC_MAJOR_4
cd->scale_vol = vol_get_processing_function(dev, cd);
#else
cd->scale_vol = vol_get_processing_function(dev, sink_c, cd);
#endif

cd->scale_vol = vol_get_processing_function(dev, sink_c);
if (!cd->scale_vol) {
comp_err(dev, "volume_prepare(): invalid cd->scale_vol");

Expand Down
133 changes: 127 additions & 6 deletions src/audio/module_adapter/module/volume/volume_generic.c
View file
Original file line number Diff line number Diff line change
Expand Up @@ -93,6 +93,47 @@ static void vol_s24_to_s24(struct processing_module *mod, struct input_stream_bu
y = audio_stream_wrap(sink, y + n);
}
}

/**
* \brief Volume passthrough from 24/32 bit to 24/32 bit.
* \param[in,out] dev Volume base component device.
* \param[in,out] sink Destination buffer.
* \param[in,out] source Input buffer.
* \param[in] frames Number of frames to process.
* \param[in] attenuation factor for peakmeter adjustment (unused)
*
* Copy and scale volume from 24/32 bit source buffer
* to 24/32 bit destination buffer.
*/
static void vol_passthrough_s24_to_s24(struct processing_module *mod,
struct input_stream_buffer *bsource,
struct output_stream_buffer *bsink, uint32_t frames,
uint32_t attenuation)
{
struct audio_stream __sparse_cache *source = bsource->data;
struct audio_stream __sparse_cache *sink = bsink->data;
int32_t *x;
int32_t *y;
int nmax, n;
const int nch = audio_stream_get_channels(source);
int remaining_samples = frames * nch;

x = audio_stream_wrap(source, (char *)audio_stream_get_rptr(source) + bsource->consumed);
y = audio_stream_wrap(sink, (char *)audio_stream_get_wptr(sink) + bsink->size);

bsource->consumed += VOL_S32_SAMPLES_TO_BYTES(remaining_samples);
bsink->size += VOL_S32_SAMPLES_TO_BYTES(remaining_samples);
while (remaining_samples) {
nmax = audio_stream_samples_without_wrap_s24(source, x);
n = MIN(remaining_samples, nmax);
nmax = audio_stream_samples_without_wrap_s24(sink, y);
n = MIN(n, nmax);
memcpy_s(y, n * sizeof(int32_t), x, n * sizeof(int32_t));
remaining_samples -= n;
x = audio_stream_wrap(source, x + n);
y = audio_stream_wrap(sink, y + n);
}
}
#endif /* CONFIG_FORMAT_S24LE */

#if CONFIG_FORMAT_S32LE
Expand Down Expand Up @@ -147,6 +188,46 @@ static void vol_s32_to_s32(struct processing_module *mod, struct input_stream_bu
y = audio_stream_wrap(sink, y + n);
}
}

/**
* \brief Volume passthrough from 32 bit to 32 bit.
* \param[in,out] dev Volume base component device.
* \param[in,out] sink Destination buffer.
* \param[in,out] source Input buffer.
* \param[in] frames Number of frames to process.
* \param[in] attenuation factor for peakmeter adjustment (unused)
*
* Copy and scale volume from 32 bit source buffer
* to 32 bit destination buffer.
*/
static void vol_passthrough_s32_to_s32(struct processing_module *mod,
struct input_stream_buffer *bsource,
struct output_stream_buffer *bsink, uint32_t frames,
uint32_t attenuation)
{
struct audio_stream __sparse_cache *source = bsource->data;
struct audio_stream __sparse_cache *sink = bsink->data;
int32_t *x;
int32_t *y;
int nmax, n;
const int nch = audio_stream_get_channels(source);
int remaining_samples = frames * nch;

x = audio_stream_wrap(source, (char *)audio_stream_get_rptr(source) + bsource->consumed);
y = audio_stream_wrap(sink, (char *)audio_stream_get_wptr(sink) + bsink->size);
bsource->consumed += VOL_S32_SAMPLES_TO_BYTES(remaining_samples);
bsink->size += VOL_S32_SAMPLES_TO_BYTES(remaining_samples);
while (remaining_samples) {
nmax = audio_stream_samples_without_wrap_s32(source, x);
n = MIN(remaining_samples, nmax);
nmax = audio_stream_samples_without_wrap_s32(sink, y);
n = MIN(n, nmax);
memcpy_s(y, n * sizeof(int32_t), x, n * sizeof(int32_t));
remaining_samples -= n;
x = audio_stream_wrap(source, x + n);
y = audio_stream_wrap(sink, y + n);
}
}
#endif /* CONFIG_FORMAT_S32LE */

#if CONFIG_FORMAT_S16LE
Expand Down Expand Up @@ -198,18 +279,58 @@ static void vol_s16_to_s16(struct processing_module *mod, struct input_stream_bu
y = audio_stream_wrap(sink, y + n);
}
}

/**
* \brief Volume passthrough from 16 bit to 16 bit.
* \param[in,out] dev Volume base component device.
* \param[in,out] sink Destination buffer.
* \param[in,out] source Input buffer.
* \param[in] frames Number of frames to process.
* \param[in] attenuation factor for peakmeter adjustment (unused)
*
* Copy and scale volume from 16 bit source buffer
* to 16 bit destination buffer.
*/
static void vol_passthrough_s16_to_s16(struct processing_module *mod,
struct input_stream_buffer *bsource,
struct output_stream_buffer *bsink, uint32_t frames,
uint32_t attenuation)
{
struct audio_stream __sparse_cache *source = bsource->data;
struct audio_stream __sparse_cache *sink = bsink->data;
int16_t *x;
int16_t *y;
int nmax, n;
const int nch = audio_stream_get_channels(source);
int remaining_samples = frames * nch;

x = audio_stream_wrap(source, (char *)audio_stream_get_rptr(source) + bsource->consumed);
y = audio_stream_wrap(sink, (char *)audio_stream_get_wptr(sink) + bsink->size);
bsource->consumed += VOL_S16_SAMPLES_TO_BYTES(remaining_samples);
bsink->size += VOL_S16_SAMPLES_TO_BYTES(remaining_samples);
while (remaining_samples) {
nmax = audio_stream_samples_without_wrap_s16(source, x);
n = MIN(remaining_samples, nmax);
nmax = audio_stream_samples_without_wrap_s16(sink, y);
n = MIN(n, nmax);
memcpy_s(y, n * sizeof(int16_t), x, n * sizeof(int16_t));
remaining_samples -= n;
x = audio_stream_wrap(source, x + n);
y = audio_stream_wrap(sink, y + n);
}
}
#endif /* CONFIG_FORMAT_S16LE */

const struct comp_func_map volume_func_map[] = {
#if CONFIG_FORMAT_S16LE
{ SOF_IPC_FRAME_S16_LE, vol_s16_to_s16 },
#endif /* CONFIG_FORMAT_S16LE */
{ SOF_IPC_FRAME_S16_LE, vol_s16_to_s16, vol_passthrough_s16_to_s16},
#endif
#if CONFIG_FORMAT_S24LE
{ SOF_IPC_FRAME_S24_4LE, vol_s24_to_s24 },
#endif /* CONFIG_FORMAT_S24LE */
{ SOF_IPC_FRAME_S24_4LE, vol_s24_to_s24, vol_passthrough_s24_to_s24},
#endif
#if CONFIG_FORMAT_S32LE
{ SOF_IPC_FRAME_S32_LE, vol_s32_to_s32 },
#endif /* CONFIG_FORMAT_S32LE */
{ SOF_IPC_FRAME_S32_LE, vol_s32_to_s32, vol_passthrough_s32_to_s32},
#endif
};

const size_t volume_func_count = ARRAY_SIZE(volume_func_map);
Expand Down
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